Communications service providers are finding it increasingly challenging to maintain consistent signal quality during a communication session as the variety of communications equipment, networks, and protocols continue to proliferate. For example, it is not uncommon for a modern communication session to be conducted between one user on a traditional landline telephone (e.g., a telephone connected to a public switched telephone network (PSTN)) and another user on personal computer hosting a voice over Internet Protocol (VoIP) session. However, even slight differences in the signal quality between the landline and VoIP connections may make it difficult for the two users to hear each other clearly during the communication session. This problem is especially acute when additional parties participate in a communication session (e.g., in a conference call) where signal quality and other audio problems can multiply accordingly.
Therefore, there is a need for an approach that provides for efficient monitoring and compensation of audio signals during a communication session.